SIP and port

We have an asterisk server where we have set up a firewall and when we activate it we can not hear the customer and the customer can not hear us.

We have for this provider opened for port 5060, does anyone have an idea about which other ports we should open for to get to the calls to work both for us and the callers?

Signaling 5060 and RTP 10000-20000 check your rtp.conf file.

The defaults for RTP are 5000 to 31000/UDP. Otherwise you need to open the range specified in rtp.conf. 10000 to 20000 is the range in the sample rtp.conf, not the default in the code.

You should not filter outbund UDP port numbers at all, as they are controlled by the remote party.

If you have NAT, you should port forward the incoming range to the Asterisk box. Some routers can snoop into SIP, but they often get things wrong.

Note you need to UDP ports for each active call leg and you also need port to cover for recently ended legs. (Odd number ports are used for RTCP.)

It appears that we were able to solve the problem based on your post. Thank you for your assistance…